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Unified Diff: talk/app/webrtc/peerconnectioninterface_unittest.cc

Issue 1403173002: Don't create remote streams if m-line direction doesn't include "send". (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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Index: talk/app/webrtc/peerconnectioninterface_unittest.cc
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 5e88658a4e25ce3c464853dfbba53a1335829c63..bee6380bf46a17275bf2bd3fb1c3e2e7ae42497c 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -71,6 +71,9 @@ static const char kStreams[][8] = {"stream1", "stream2"};
static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
+static const char kRecvonly[] = "recvonly";
+static const char kSendrecv[] = "sendrecv";
+
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1" and a video track with id "video_1;
static const char kSdpStringWithStream1[] =
@@ -84,12 +87,14 @@ static const char kSdpStringWithStream1[] =
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
@@ -110,6 +115,7 @@ static const char kSdpStringWithStream1And2[] =
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
@@ -117,6 +123,7 @@ static const char kSdpStringWithStream1And2[] =
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
@@ -135,9 +142,11 @@ static const char kSdpStringWithoutStreams[] =
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams. Msid is supported.
@@ -153,9 +162,11 @@ static const char kSdpStringWithMsidWithoutStreams[] =
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams and audio only.
@@ -170,6 +181,7 @@ static const char kSdpStringWithoutStreamsAudioOnly[] =
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
@@ -184,10 +196,12 @@ static const char kSdpStringSendOnlyWithoutStreams[] =
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtpmap:120 VP8/90000\r\n";
@@ -205,11 +219,13 @@ static const char kSdpStringInit[] =
static const char kSdpStringAudio[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
+ "a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringMs1Audio0[] =
@@ -1783,6 +1799,23 @@ TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
// No crash is a pass.
}
+// This tests that if a recvonly remote description is set, no remote streams
+// will be created, even if the description contains SSRCs/MSIDs.
+// See: https://code.google.com/p/webrtc/issues/detail?id=5054
+TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
+ FakeConstraints constraints;
+ constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+ true);
+ CreatePeerConnection(&constraints);
+
+ std::string recvonly_offer = kSdpStringWithStream1;
+ rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
+ strlen(kRecvonly), &recvonly_offer);
+ CreateAndSetRemoteOffer(recvonly_offer);
+
+ EXPECT_EQ(0u, observer_.remote_streams()->count());
+}
+
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and no MSID support.
// It also tests that the default stream is updated if a video m-line is added
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